Support SIP_CODEC channel variable for early media. (Imported from 1.2, with a small
authorOlle Johansson <oej@edvina.net>
Thu, 9 Mar 2006 17:41:38 +0000 (17:41 +0000)
committerOlle Johansson <oej@edvina.net>
Thu, 9 Mar 2006 17:41:38 +0000 (17:41 +0000)
change for const char* channel variables)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12478 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index 06ef739..7e1f828 100644 (file)
@@ -2605,12 +2605,34 @@ static int sip_hangup(struct ast_channel *ast)
        return 0;
 }
 
+/*! \brief Try setting codec suggested by the SIP_CODEC channel variable */
+static void try_suggested_sip_codec(struct sip_pvt *p)
+{
+       int fmt;
+       const char *codec;
+
+       codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC");
+       if (!codec) 
+               return;
+
+       fmt = ast_getformatbyname(codec);
+       if (fmt) {
+               ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC) variable\n", codec);
+               if (p->jointcapability & fmt) {
+                       p->jointcapability &= fmt;
+                       p->capability &= fmt;
+               } else
+                       ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n");
+       } else
+               ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n", codec);
+       return; 
+}
+
 /*! \brief  sip_answer: Answer SIP call , send 200 OK on Invite 
  * Part of PBX interface */
 static int sip_answer(struct ast_channel *ast)
 {
-       int res = 0,fmt;
-       const char *codec;
+       int res = 0;
        struct sip_pvt *p = ast->tech_pvt;
 
        ast_mutex_lock(&p->lock);
@@ -2618,19 +2640,7 @@ static int sip_answer(struct ast_channel *ast)
 #ifdef OSP_SUPPORT     
                time(&p->ospstart);
 #endif
-       
-               codec=pbx_builtin_getvar_helper(p->owner,"SIP_CODEC");
-               if (codec) {
-                       fmt=ast_getformatbyname(codec);
-                       if (fmt) {
-                               ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC) variable\n",codec);
-                               if (p->jointcapability & fmt) {
-                                       p->jointcapability &= fmt;
-                                       p->capability &= fmt;
-                               } else
-                                       ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n");
-                       } else ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n",codec);
-               }
+               try_suggested_sip_codec(p);     
 
                ast_setstate(ast, AST_STATE_UP);
                if (option_debug)
@@ -4671,6 +4681,7 @@ static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_r
        }
        respprep(&resp, p, msg, req);
        if (p->rtp) {
+               try_suggested_sip_codec(p);     
                add_sdp(&resp, p);
        } else {
                ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);